Abstract: A voice enhancement logic improves the perceptual quality of a processed voice. The voice enhancement system includes a passing tire hiss noise detector and a passing tire hiss noise attenuator. The passing tire hiss noise detector detects a passing tire hiss noise by modeling the passing tire hiss. The passing tire hiss noise attenuator dampens the passing tire hiss noise to improve the intelligibility of a speech signal.

Abstract: The invention relates to a method of automatic sound recognition. The object of the present invention is to provide an alternative scheme for automatically recognizing sounds, e.g. human speech. The problem is solved by providing a training database comprising a number of models, each model representing a sound element in the form of a binary mask comprising binary time frequency (TF) units which indicate the energetic areas in time and frequency of the sound element in question, or of characteristic features or statistics extracted from the binary mask; providing an input signal comprising an input sound element; estimating the input sound element based on the models of the training database to provide an output sound element. The method has the advantage of being relatively simple and adaptable to the application in question. The invention may e.g. be used in devices comprising automatic sound recognition, e.g. for sound, e.g. voice control of a device, or in listening devices, e.g.

Abstract: A speech signal processing apparatus includes a control signal output unit configured to receive as an input signal either one of a first speech signal corresponding to a sound uttered by a user and a second speech signal corresponding to a sound output from an eardrum of the user when the user utters a sound, and output a control signal corresponding to a noise level of the input signal, and a speech signal output unit configured to output either one of the first speech signal and the second speech signal according to the control signal.

Abstract: The invention relates to a compressor and method for amplifying an input signal with a controlled gain. An output signal representing the input signal is amplified by an initial gain and a signal level of the input signal or of the output signal is compared with a threshold level. If the signal level is below the threshold level, the initial gain value is updated using an adaptive control characteristic, and if the signal level is above the threshold level, the initial gain value is updated using a fixed control characteristic or an adaptive control characteristic respectively. The adaptive control characteristic is dependent on the signal level and the fixed control characteristic is independent from the signal level.

Abstract: The APPARATUSES, METHODS AND SYSTEMS FOR SPARSE SINUSOIDAL AUDIO PROCESSING AND TRANSMISSION (hereinafter “SS-Audio”) provides a platform for encoding and decoding audio signals based on a sparse sinusoidal structure. In one embodiment, the SS-Audio encoder may encode received audio inputs based on its sparse representation in the frequency domain and transmit the encoded and quantized bit streams. In one embodiment, the SS-Audio decoder may decode received quantized bit streams based on sparse reconstruction and recover the original audio input by reconstructing the sinusoidal parameters in the frequency domain.

Abstract: The voice clarification apparatus includes a plurality of band-pass filters that respectively extract a plurality of band components, which are included in a voice band, from an input audio signal; a gain determination unit that determines a gain according to the level of a signal of a band component which is extracted by at least one band-pass filter of the plurality of band-pass filters; a level adjustment unit that adjusts the levels of signals of the plurality of band components which are extracted by the plurality of band-pass filters using the gain; and a first addition unit that adds a signal which is based on the audio signal to a signal in which the gain is adjusted by the level adjustment unit, and outputs a signal obtained through the addition.

Abstract: A coded code string from an input terminal 110 is demultiplexed by a demultiplexer circuit 101, normalization coefficient information in the code string is sent to a normalization coefficient information increasing/decreasing circuit 102, addition or subtraction of a positive value is performed, and level adjustment of a signal is performed. A normalization coefficient information cutoff amount calculating circuit 103 calculates the cutoff amount for a case where the subtraction amount of normalization coefficient information is larger than normalization coefficient information and normalization coefficient information after subtraction is cut off at the minimum possible value. A gain control function generation information modifying circuit 104 modifies gain control function generation information according to the cutoff amount.

Abstract: A sound processing apparatus is provided for estimating the power of background noise using a directional sound receiving technology using a plurality of sound receiving units, computing a gain control value on the basis of the estimated power of background noise and a predetermined power target value, and outputting the gain control value, so that a delay time of starting gain control can be reduced, and a slow response of a speech recognition application program or degradation of the speech quality of a voice communication program can be prevented.

Abstract: A device for continuous time quantization of an input signal, in order to supply a continuous time output signal that is quantized as two bits, the device including: an electronic circuit, designed to supply a first bit of the output signal called the sign bit which at any time takes a first value when the input signal is positive and a second value when the input signal is negative, and an envelope analysis circuit designed to supply a second bit of the output signal called the envelope variation bit which at any time takes a first value, called high value, when an envelope signal of the input signal is increasing, and a second value, called low value, when the envelope signal is decreasing.

Abstract: An audio encoding method and a corresponding decoding method are provided. Accordingly, the pre-echo effect of the audio transient signal is eliminated and the distortion of the transient signal is mitigated. The technical solution includes performing time-domain processing on an input audio transient signal; dividing sampling points x1,x2, . . . , xN of an input frame into L segments; calculating an energy Ei for each segment; calculating an average energy E0 for each segment of the input frame; calculating a multiplying parameter ?i corresponding to each segment by virtue of ?i=r(bitrate)*E0/Ei; multiplying the sampling points of all the segments of the input frame by corresponding multiplying parameter ?i, obtaining the processed sampling points x1?,x2?, . . . , xN?; and sending the multiplying parameter ?i to a code stream for transportation; performing time-frequency transformation and coding on the processed sampling points x1?,x2?, . . . , xN? and outputting to the code stream.

Abstract: There is disclosed an encoding device capable of improving similarity between the high frequency band spectrum of the original signal and a new spectrum to be generated while realizing a low bit rate when encoding a wide-band signal spectrum. The encoding device has sub-band amplitude calculation units (122, 128) for calculating the amplitude of the respective sub-bands for the high frequency band spectrum obtained from the wide-band signal. A search unit (124) and a gain codebook (125) select some sub-bands from a plurality of sub-bands and only the gain of the selected sub-bands is subjected to encoding. An interpolation unit (126) expresses the gain of the sub-band not selected, by mutually interpolating the selected gains.

Abstract: An apparatus comprising at least one processor and at least one memory including computer program code. The at least one memory and the computer program code is configured to, with the at least one processor, cause the apparatus at least to determine a loudness estimate of a first audio signal, generate a parameter dependent on the loudness estimate; and control the first audio signal dependent on the parameter.

Abstract: In a method of smoothing background noise in a telecommunication speech session; receiving and decoding S1O a signal representative of a speech session, the signal comprising both a speech component and a background noise component. Subsequently, determining LPC parameters S20 and an excitation signal S30 for the received signal. Thereafter, synthesizing and outputting (S40) an output signal based on the determined LPC parameters and excitation signal. In addition, modifying S35 the determined excitation signal by reducing power and spectral fluctuations of the excitation signal to provide a smoothed output signal.

Abstract: A method for performing a call between a near-end user and a far-end user, which includes the following operations performed during the call by the near-end user's communications device. Automatic gain control (AGC) is performed to update a gain applied to an uplink speech signal. A frame is detected in a downlink signal that contains speech; in response, the updating of the gain is frozen. Other embodiments are also described and claimed.

Abstract: Disclosed is a decoding device which can efficiently encode/decode spectral data in a high pass section of a broadband signal. In the disclosed device: a sample group extraction unit (372) partially selects spectral components by means of an ease of selection importance which is the extent that the spectral components come close to the spectral component having the maximum amplitude value, in the spectrum of a high pass estimated by means of first parameters contained in second encoded information and bands most approximated to each of the spectrums of a plurality of sub-bands calculated from the spectrum of a second decode signal; a logarithmic gain application unit (373) applies second parameters to the partially selected spectral components; and an interpolation processing unit (374) applies third parameters which are adaptively set according to the value of the second parameters, to the spectral components which were not partially selected.

Abstract: A method and apparatus are provided for generating a noise reduced output signal from sound received by a first microphone. The method includes transforming the sound received by the first microphone into a first input signal and transforming sound received by a second microphone into a second input signal. The method includes calculating, for each of a plurality of frequency components, an energy transfer function value as a real-valued quotient by dividing a temporally averaged product of an amplitude of the first input signal and the second input signal by a temporally averaged absolute square of the second input signal, calculating a gain value as a function of the calculated energy transfer function value, and generating the noise reduced output signal based on the product of the first input signal and the calculated gain value at each of the plurality of frequency components.

Abstract: The invention provides a noise removal device and method capable of more proper interpolation on an input signal. The noise removal device is for removing noise in an input signal and includes: a noise detector detecting noise in an IF signal and outputting a noise detection signal; an interpolation controller determining a period and amount of interpolation for noise correction processing, based on the IF signal and the noise detection signal; and a noise gate processor performing the noise correction processing on the IF signal, based on the interpolation period and amount supplied from the interpolation controller. The interpolation controller sets a predetermined first interpolation period, based on a first noise detection signal inputted from the noise detector, and redefines a second interpolation period longer than the first interpolation period when a second noise detection signal is detected within the first interpolation period.

Abstract: A noisy signal is picked up by a microphone, digitized by an Analog to Digital Converter and fed to a processor for analysis and wind noise reduction. Most of noise reduction methods are based on the assumption that the interfering noise is stationary or slowly varying compared with speech. This assumption allows “learning” the characteristics of the noise between speech pauses and, based on a noise estimate, to build different filters that reduce the noise. In the case of wind noise this basic assumption is not valid. Wind noise is highly non-stationary, its power and spectral characteristics vary greatly. Because wind noise is not stationary, regular noise reduction methods cannot be used to reduce wind noise. For reducing wind noise effects in a device, the presence of wind should be detected reliably and then a novel approach presented here must be applied to eliminate the wind noise.

Abstract: An apparatus for processing an input sound signal, the apparatus including: gain circuitry configured to control a gain based on a plurality of respective sub-signals of the input sound signal; and an amplification apparatus configured to adjust the amplification of all the plurality of amplitudes based on the gain.

Abstract: The voice quality optimization system includes a controller that controls voice quality by adjusting parameters that control voice quality characteristics of the communication device; and a measuring unit that measures voice quality of the communication device and transmits the measured voice quality as a feedback to the controller. The controller controls voice quality by calibrating the parameters of the communication device, including a receiving sensitivity/frequency response characteristic curve, receiving loudness rating and idle channel noise-receiving. A method for setting voice optimization in a communication device includes measuring parameters of the communication device, determining whether the parameters of the communication device are within a target range, and calibrating a first parameter to be within the target range if the first parameter is outside the target range.

Abstract: Provided are a method and apparatus for encoding and decoding a high frequency signal by using a low frequency signal. The high frequency signal can be encoded by extracting a coefficient by linear predicting a high frequency signal, and encoding the coefficient, generating a signal by using the extracted coefficient and a low frequency signal, and encoding the high frequency signal by calculating a ratio between the high frequency signal and an energy value of the generated signal. Also, the high frequency signal can be decoded by decoding a coefficient, which is extracted by linear predicting a high frequency signal, and a low frequency signal, and generating a signal by using the decoded coefficient and the decoded low frequency signal, and adjusting the generated signal by decoding a ratio between the generated signal and an energy value of the high frequency signal.

Abstract: A voice recognition system comprises: a voice input unit that receives an input signal from a voice input element and output it; a voice detection unit that detects an utterance segment in the input signal; a voice recognition unit that performs voice recognition for the utterance segment; and a control unit that outputs a control signal to at least one of the voice input unit and the voice detection unit and suppresses a detection frequency if the detection frequency satisfies a predetermined condition.

Abstract: A method for processing signals for reducing noise components in a vehicular microphone system (500A) compares (517) a plurality of noise component values (511, 513, 515) for at least one frequency band of plurality of frequency bands. At least one frequency band is then downwardly expanded (521) by a predetermined expansion ratio (523) for providing a noise reduced signal (535) when determining a value is below a predetermined threshold (519).

Abstract: The present invention relates to signal modification before pitch period repetition for the synthesis of blocks lost on decoding digital audio signals. The effects of repetition of transitories, such as the plosives of a speech signal, are avoided by comparing the samples of a pitch period with those of the previous pitch period. The signal is modified preferentially by taking the minimum between a current sample (e(3)) of the last pitch period (Tj) and at least one sample (e(2?T0) of approximately the same position in the previous pitch period (Tj?1).

Abstract: Disclosed is a decoder capable of improving the sound quality of a decoded sound signal in an encoding method which combines speech encoding and music encoding in a hierarchical structure. A transform-encoding decoding unit (202) decodes transform-encoded data to generate a spectrum of a decoded transform-encoded signal. A band decision unit (203) uses the spectrum of the decoded transform-encoded signal to decide whether each of a plurality of bands in which frequency components of an input signal are divided constitute a first band in which a transform encoded pulse is not established or a second band in which said pulse is established. A CELP component suppression unit (207) suppresses the spectrum of a CELP decoded signal, which is the frequency component of a decoded signal of CELP encoded data, to the extent that suppression in the first band is weaker than suppression in the second band.

Abstract: Disclosed is a coding apparatus and method using residual bits. Accordingly, performance (voice quality) is enhanced by quantizing a full-band gain of frequency coefficients existing in sub-bands to which bits are not assigned in an algebraic vector quantization (AVQ). Further, the performance (voice quality) is enhanced by sequentially quantizing a sub-band gain of sub-bands to which bits are not assigned until residual bits are removed. Furthermore, the performance (voice quality) is enhanced by demodulating AVQ coefficients, and correcting quantization noises starting with a coefficient having the greatest absolute coefficient among the AVQ coefficients, when residual bits additionally remain.

Abstract: An apparatus and a method for automatically controlling a gain using phase information are provided. The apparatus includes a frequency conversion unit converting each of input signals received from a plurality of acoustic input apparatuses to frequency input signals, a factor determination unit determining a scaling factor according to a gain difference between the input signals based on a phase difference of the frequency input signals, and a scaling performance unit performing scaling for gain compensation between the input signals based on the scaling factor.

Abstract: A level of ambient noise at a local device is determined. A dynamic range compression (DRC) gain is computed based on the level of ambient noise at the local device. An additional gain factor is computed. A total gain is computed based on an adding of the DRC gain and the additional gain factor. An amplitude of an audio signal is adjusted based on the total gain, wherein the audio signal was transmitted from a remote device and received by the local device.

Abstract: An apparatus and method of improving the quality of a speech codec are provided. In the method, a first energy of a signal decoded by a low-band codec is calculated, and a second energy of a signal decoded by a low-band enhancement mode is calculated. Then, when the first energy is less than a first threshold value or less than a product of the second energy and a second threshold value, a size of the decoded signal is scaled. Accordingly, generation of a quantization error with respect to a silence segment is reduced.

Abstract: A voice-activated command and control system for remotely-controlled vehicles includes a voice-activated control module, a microphone, and a verbal or visual feedback indicator such as a speaker. The operator speaks instructions into the microphone to activate control functions on the vehicle. Confirmation of receipt and/or status of the verbally commanded instructions are sent back to the operator such as spoken through the speaker. The microphone and speaker can be implemented, for example, either in a headset (with a microphone and earphones) worn by the operator, or in the hand-held controller for the vehicle. The system can be alternately implemented without the speaker or other feedback element.

Abstract: Disclosed is a gain control system in which speech model constituted from a sound pressure and a feature is stored in a speech model storage unit for each of a plurality of phonemes or for each of clusters into which a speech is divided. When an input signal is given, a feature conversion unit calculates a feature and a sound pressure of the input signal. A sound pressure comparison unit determines a sound pressure ratio between the input signal and each of speech models. A distance calculation unit calculates a distance between the feature of the input signal and the feature of each of the speech models. A gain calculation unit calculates a gain value from the sound pressure ratio and information on the distance. A sound pressure compensation unit thereby compensates for the sound pressure of the input signal.

Abstract: An apparatus and method of improving the quality of a speech codec are provided. In the method, a first energy of a signal decoded by a low-band codec is calculated, and a second energy of a signal decoded by a low-band enhancement mode is calculated. Then, when the first energy is less than a first threshold value or less than a product of the second energy and a second threshold value, a size of the decoded signal is scaled. Accordingly, generation of a quantization error with respect to a silence segment is reduced.

Abstract: A clipping detection device calculates an amplitude distribution of an input signal for each predetermined period, calculates a deflection degree of the distribution on the basis of the calculated amplitude distribution, and then detects clipping of a communication signal on the basis of the calculated deflection degree of the distribution.

Abstract: In general, the techniques are described for adjusting audio gain levels for multi-talker audio. In one example, an audio system monitors an audio stream for the presence of a new talker. Upon identifying a new talker, the system determines whether the new talker is a first-time talker. For a first-time talker, the system executes a fast-attack/decay automatic gain control (AGC) algorithm to quickly determine a gain value for the first-time talker. The system additionally executes standard AGC techniques to refine the gain for the first-time talker while the first-time talker continues speaking. When a steady state within a decibel threshold is attained using standard AGC for the first-time talker, the system stores the steady state gain for the first-time talker to storage. Upon identifying a previously-identified talker, the system retrieves from storage the steady state gain for the talker and applies the steady state gain to the audio stream.

Abstract: A method is provided for concealing a transmission error in a digital signal chopped into a plurality of successive frames associated with different time intervals in which, on reception, the signal may comprise erased frames and valid frames, the valid frames comprising information relating to the concealment of frame loss. The method is implemented during a hierarchical decoding using a core decoding and a transform-based decoding using windows introducing a time delay of less than a frame with respect to the core decoding. The method includes concealing a first set of missing samples for the erased frame, implemented in a first time interval; a step of concealing a second set of missing samples utilizing information of said valid frame and implemented in a second time interval; and a step of transition between the first and the second set of missing samples to obtain at least part of the missing frame.

Abstract: A system for coding a hierarchical audio signal, comprising, at least, a core layer using parametric coding by analysis by synthesis in a first frequency band, a band extension layer for widening said first frequency band into a second frequency band, or wideband. The system also comprises a wideband audio coding quality enhancement layer based on transform coding using a spectral parameter obtained from said band extension layer. Application to transmitting speech and/or audio signals over packet networks.

Abstract: Acoustic echo control for hands-free phone has acoustic echo cancellation and echo suppression with a voice activity detection for the echo suppression based on near-end input power together with an estimate for acoustic echo cancellation gain.

Abstract: Provided is an apparatus and method for coding and decoding multi-object audio signals with various channels and providing backward compatibility with a conventional spatial audio coding (SAC) bitstream. The apparatus includes: an audio object coding unit for coding audio-object signals inputted to the coding apparatus based on a spatial cue and creating rendering information for the coded audio-object signals, where the rendering information provides a coding apparatus including spatial cue information for audio-object signals; channel information of the audio-object signals; and identification information of the audio-object signals, and used in coding and decoding of the audio signals.

Abstract: Systems and methods for operating a telecommunications device in whisper mode are presented. The method generally includes adjusting a transmit audio signal gain responsive to a transmit audio signal voice activity status, transmit audio signal speech level, transmit audio signal signal-to-noise ratio, and receive audio signal voice activity status. A sidetone feedback signal gain is adjusted in conjunction with adjusting the transmit audio signal gain.

Abstract: A method (400, 500) and apparatus (220) seeks to improve the intelligibility of speech emitted into a noisy environment. Formants are identified (426) and perceptual frequency scale band is selected (502) that includes at least one of the identified formants. The SNR in each band is compared (504) to a threshold and, if the SNR for that band is less than the threshold, the method increases a formant enhancement gain for that band. A set of high pass filter gains (338) is combined (516) with the formant enhancement gains yielding combined gains that are then clipped (518), scaled (520) according to a total SNR, normalized (526), smoothed across time (530) and frequency (532), and used to reconstruct (532, 534) an audio signal.

Abstract: A speech signal is received at an input. At least one electrical value associated with the received speech signal is tracked. A dynamic adjustment of the speech signal is determined. The dynamic adjustment is selected at least in part so as to minimize a distortion and minimize an over-amplification of the speech signal based at least in part upon an analysis of the at least one electrical value. The dynamic adjustment is further selected to obtain a desired output signal characteristic for the speech signal presented at an output. The dynamic adjustment value is applied to the speech signal and the adjusted speech signal is presented at the output. The gain of the signal can also be limited to prevent over-amplification.

Abstract: An apparatus for performing improved audio processing may include a processor. The processor may be configured to divide respective signals of each channel of a multi-channel audio input signal into one or more spectral bands corresponding to respective analysis frames, select a leading channel from among channels of the multi-channel audio input signal for at least one spectral band, determine a time shift value for at least one spectral band of at least one channel, and time align the channels based at least in part on the time shift value.

Abstract: A system for controlling dynamic range of an audio signal comprises an automatic gain control element that receives an input signal having a varying level and outputs a control signal that varies based on the varying level of the input signal and a modified input signal having a dynamic range different than a dynamic range of the input signal. The system also comprises an inverter that inverts the control signal or a block-based control signal corresponding to the control signal in block format. The system also comprises a variable gain element that receives the modified input signal and at least some of the inverted control signal or block-based control signal. The variable gain element also outputs a remainder signal corresponding to the modified input signal as unmodified based on the at least some of the inverted control signal or block-based control signal.

Abstract: An audio input device is provided which can include a number of features. In some embodiments, the audio input device includes a housing, a microphone carried by the housing, and a processor carried by the housing and configured to modify an input sound signal so as to amplify frequencies corresponding to a target human voice and diminish frequencies not corresponding to the target human voice. In another embodiment, an audio input device is configured to treat an auditory gap condition of a user by extending gaps in continuous speech and outputting the modified speech to the user. In another embodiment, the audio input device is configured to treat a dichotic hearing condition of a user. Methods of use are also described.

Abstract: This invention proposes a more efficient way to quantize temporal envelope shaping of high band signal by benefiting from energy relationship between low band signal and high band signal; if low band signal is well coded or it is coded with time domain codec such as CELP, temporal envelope shaping information of low band signal can be used to predict temporal envelope shaping of high band signal; the temporal envelope shaping prediction can bring significant saving of bits to precisely quantize temporal envelope shaping of high band signal. This prediction approach can be combined with other specific approach to further increase the efficiency and save mores bits.

Abstract: An encoding device improves the sound quality of a stereo signal while maintaining a low bit rate. The encoding device includes: an LP inverse filter which LP-inverse-filters a left signal L(n) by using an inverse quantization linear prediction coefficient AdM(z) of a monaural signal; a T/F conversion unit which converts the left sound source signal Le(n) from a temporal region to a frequency region; an inverse quantizer which inverse-quantizes encoded information Mqe; spectrum division units which divide a high-frequency component of the sound source signal Mde(f) and the left signal Le(f) into a plurality of bands; and scale factor calculation units which calculate scale factors ai and ssi by using a monaural sound source signal Mdeh,i(f), a left sound source signal Leh,i(f), Mdeh,i(f), and right sound source signal Reh,i(f) of each divided band.

Abstract: An audio input signal is filtered using an adaptive filter to generate a prediction output signal with reduced noise, wherein the filter is implemented using a plurality of coefficients to generate a plurality of prediction errors and to generate an error from the plurality of prediction errors, wherein the absolute values of the coefficients are continuously reduced by a plurality of reduction parameters.

Abstract: A method and system for filtering a multi-channel audio signal having a speech channel and at least one non-speech channel, to improve intelligibility of speech determined by the signal. In typical embodiments, the method includes steps of determining at least one attenuation control value indicative of a measure of similarity between speech-related content determined by the speech channel and speech-related content determined by the non-speech channel, and attenuating the non-speech channel in response to the at least one attenuation control value. Typically, the attenuating step includes scaling of a raw attenuation control signal (e.g., a ducking gain control signal) for the non-speech channel in response to the at least one attenuation control value. Some embodiments are a general or special purpose processor programmed with software or firmware and/or otherwise configured to perform filtering in accordance the invention.

Abstract: A voice data transferring device intermediates between an in-vehicle terminal and a voice recognition server. In order to check a change in voice recognition performance of the voice recognition server, the voice data transferring device performs a noise suppression processing on a voice data for evaluation in a noise suppression module; transmits the voice data for evaluation to the voice recognition server; and receives a recognition result thereof. The voice data transferring device sets a value of a noise suppression parameter used for a noise suppression processing or a value of a result integration parameter used for a processing of integrating a plurality of recognition results acquired from the voice recognition server, at an optimum value, based on the recognition result of the voice recognition server. This makes it possible to set a suitable parameter even if the voice recognition performance of the voice recognition server changes.

Abstract: Mechanisms are known that allow receivers to control loudness of speech in broadcast signals but these mechanisms require an estimate of speech loudness be inserted into the signal. Disclosed techniques provide improved estimates of loudness. According to one implementation, an indication of the loudness of an audio signal containing speech and other types of audio material is obtained by classifying segments of audio information as either speech or non-speech. The loudness of the speech segments is estimated and this estimate is used to derive the indication of loudness. The indication of loudness maybe used to control audio signal levels so that variations in loudness of speech between different programs is reduced. A preferred method for classifying speech segments is described.